1. Field of the Invention
The present invention generally relates to teleconferencing, and in particular, to a system and method that improves audio quality by compensating for lossy audio compression algorithms used in conventional conferencing networks.
2. Description of the Related Art
Computer-based conferencing has become popular as a means for carrying out a meeting between remotely located participants. As shown in FIG. 1, a conventional conferencing network 10 typically includes a plurality of clients 14-16 connected to a conference server 12 via a communications network 15. The server 12 permits the clients 14-16 to simultaneously communicate with one another, effectively emulating a conference room filled with participants.
Although many computer-based conferencing systems permit transfer of video, voice, and data, the system 10 shown in FIG. 1 illustrates only the audio portion of the conferencing system. In the system 10, the clients 14-16 can be any device permitting a user to transmit and receive compressed audio data, such as a personal computer (PC), a video phone, a telephone featuring audio compression, or the like. Each of the clients 14-16 includes an audio compression algorithm 18-20 and a corresponding decompression algorithm 22-24. When a participant speaks, his/her utterances are converted to digital information by the respective clients 14-16. This digital information is then compressed by the compression algorithms 18-20 and transferred over the network 15 to the conference server 12.
To maintain a fully duplexed audio channel, the conference server 12 separately decompresses each of the compressed audio streams received from the clients 14-16 using the decompressor 26. The decompressed audio channels are then summed by a summation circuit 28. The output of the summation circuit is then compressed by a compressor 30. Representing the fully duplexed channel, the compressed summation stream is then transmitted back to each of the clients 14-16 over the network 15. Each of the clients 14-16 includes a decompression algorithm 22-24 which decompresses the audio information carried in the common channel.
In recent years, there has been major progress in the development of compression algorithms. This represents a real economic benefit, since compressed voice takes much less bandwidth than uncompressed voice and many conversations can be multiplexed over the same channel. However, compression algorithms such as GSM, True Speech.TM., G.723.1, etc., are lossy and hence, pose a problem for features like conference calling. Using lossy algorithms results in data loss, and consequently introduces distortism every time the audio is compressed. This problem becomes more compounded in hierarchical networks, where summation streams from various servers are successively combined using lossy compression algorithms. Every round of compression/decompression results in a poorer voice quality.
Alternatively, lossless compression algorithms can be used to overcome the problems caused by repeated transformations. However, lossless algorithms have lower compression ratios than lossy algorithms and therefore consume much more bandwidth. In addition, lossy compression is essential in narrow band communication systems, such as systems using H.324.
Therefore, there is a need for an improved conferencing system that does not rely on lossless audio algorithms and compensates for degraded audio quality caused by lossy compression transformations.